Hi Carsten, no is not about just rewriting the SDP. i need my UACs media to relay on my rtpproxy currently my UACs are sending the media to a private ip. my rtpproxy is in behind nat and UACs behind another nat.
On Wed, Jul 6, 2011 at 3:15 PM, Carsten Bock <cars...@ng-voice.com> wrote: > Hi MingHon, > > what do you want to achieve? If it is only about rewritibng the SDP, > then this will help you: > > fix_nated_sdp("10", "<your-ip-here>"); > => 0x02 rewrite media IP address (c=) with the provided IP address > => 0x08 rewrite IP from origin description (o=) with the provided IP > address > > Kind regards, > Carsten > > 2011/7/6 MingHon <gming...@gmail.com>: > > hello List, > > anyone could give some hints?? > > im still unable to rewrite the sdp body. > > hope to hear from you all. > > thanks > > -- > > Regards, > > > > MingHon > > > > > > On Tue, Jul 5, 2011 at 3:49 PM, MingHon <gming...@gmail.com> wrote: > >> > >> Hi List, > >> im facing an issue that my kamailio proxy did not replace the ip address > >> in the invite and 200OK sdp body. > >> my rtpproxy is running: rtpproxy -l 192.168.1.3 -u:*:7722 -u user > >> my kamailio is listening on 192.168.1.3, also > >> define: advertised_address="175.136.223.112"; & advertised_port=5060; > >> and my asterisk is on 192.168.1.23. > >> sip signalling and rtp port forwarded to kamailio. > >> uacs from another nat register successfully. > >> if i put 2 lines of force_rtp_proxy("fcow","175.136.223.112"); > >> i will get double ip addr in c and o but kamailio ignore my ip addr. > >> example i will get > >> c=IN IP4 192.168.1.3192.168.1.3 > >> here is part of my simple script. > >> hope you can help. > >> thank you very much. > >> ---------------cfg------------------- > >> route[RTPPROXY] { > >> #!ifdef WITH_NAT > >> if (is_method("BYE")) { > >> unforce_rtp_proxy(); > >> } else if (is_method("INVITE")){ > >> force_rtp_proxy("fcow","175.136.223.112"); > >> #force_rtp_proxy("fcow","175.136.223.112"); > >> xlog("L_INFO","offer"); > >> } > >> if (!has_totag()) add_rr_param(";nat=yes"); > >> #!endif > >> return; > >> } > >> -------------------------------------- > >> and here is the wireshark for uac INVITE and OK. > >> -----------INVITE----------------- > >> ve0 > >> EE;p9INVITE sip:102@192.168.2.132:5062 SIP/2.0 > >> Record-Route: <sip:192.168.1.3;lr=on;ftag=as032358a3;nat=yes> > >> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK09d5.c5e9e8d2.0 > >> Via: SIP/2.0/UDP 192.168.1.23:5080;branch=z9hG4bK71c27189;rport=5080 > >> Max-Forwards: 69 > >> From: "101" <sip:1...@aextddns.dyndns.info>;tag=as032358a3 > >> To: <sip:102@192.168.1.3:5060> > >> Contact: <sip:102@192.168.1.23:5080> > >> Call-ID: 416f6e09674ae9671bb7144a1cb11...@aextddns.dyndns.info > >> CSeq: 102 INVITE > >> User-Agent: Asterisk PBX 1.6.2.18 > >> Date: Tue, 05 Jul 2011 07:20:53 GMT > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > >> Supported: replaces, timer > >> Content-Type: application/sdp > >> Content-Length: 327 > >> v=0 > >> o=root 1639709788 1639709788 IN IP4 192.168.1.3 > >> s=Asterisk PBX 1.6.2.18 > >> c=IN IP4 192.168.1.3 > >> t=0 0 > >> m=audio 10072 RTP/AVP 0 3 8 101 > >> a=rtpmap:0 PCMU/8000 > >> a=rtpmap:3 GSM/8000 > >> a=rtpmap:8 PCMA/8000 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16 > >> a=silenceSupp:off - - - - > >> a=ptime:20 > >> a=sendrecv > >> a=nortpproxy:yes > >> -----------200OK--------------- > >> e90 > >> ElE;pX4tSIP/2.0 200 OK > >> Via: SIP/2.0/UDP > >> 192.168.2.200:5062 > ;rport=2788;received=175.138.21.31;branch=z9hG4bK2086380416 > >> Record-Route: <sip:192.168.1.3;lr=on;ftag=1796959074;nat=yes> > >> From: "101" <sip:1...@aextddns.dyndns.info>;tag=1796959074 > >> To: <sip:1...@aextddns.dyndns.info>;tag=as2e4c0125 > >> Call-ID: 1985782590@192.168.2.200 > >> CSeq: 21 INVITE > >> Server: Asterisk PBX 1.6.2.18 > >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > >> Supported: replaces, timer > >> Contact: <sip:102@192.168.1.23:5080> > >> Content-Type: application/sdp > >> Content-Length: 286 > >> v=0 > >> o=root 403900934 403900934 IN IP4 192.168.1.23 > >> s=Asterisk PBX 1.6.2.18 > >> c=IN IP4 192.168.1.23 > >> t=0 0 > >> m=audio 14420 RTP/AVP 0 8 101 > >> a=rtpmap:0 PCMU/8000 > >> a=rtpmap:8 PCMA/8000 > >> a=rtpmap:101 telephone-event/8000 > >> a=fmtp:101 0-16 > >> a=silenceSupp:off - - - - > >> a=ptime:20 > >> a=sendrecv > >> ------------------------------------ > >> My kamailio log. > >> -----------LOG------------------ > >> DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found > valid > >> DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10070 192.168.1.3 > >> INFO: <script>: offer > >> ------------------------------------- > >> double force_rtp_proxy > >> --------kamailio -> asterisk [INVITE]--------- > >> Pyi-}E7V@:#pINVITE sip:1...@aextddns.dyndns.info SIP/2.0 > >> Record-Route: <sip:192.168.1.3;lr=on;ftag=640933430;nat=yes> > >> Via: SIP/2.0/UDP 192.168.1.3;branch=z9hG4bK89a5.53e9f766.0 > >> Via: SIP/2.0/UDP > >> 192.168.2.200:5062 > ;rport=2788;received=175.138.21.31;branch=z9hG4bK1673765648 > >> From: "101" <sip:1...@aextddns.dyndns.info>;tag=640933430 > >> To: <sip:1...@aextddns.dyndns.info> > >> Call-ID: 1909950509@192.168.2.200 > >> CSeq: 21 INVITE > >> Contact: <sip:101@175.138.21.31:2788> > >> Content-Type: application/sdp > >> Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, > >> SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE > >> Max-Forwards: 69 > >> User-Agent: T20 9.41.0.80 > >> Allow-Events: talk,hold,conference,refer,check-sync > >> Content-Length: 334 > >> v=0 > >> o=20073 20073 IN IP4 192.168.1.3192.168.1.3 > >> s=SDP data > >> c=IN IP4 192.168.1.3192.168.1.3 > >> t=0 0 > >> m=audio 1006410064 RTP/AVP 0 8 18 9 101 > >> a=rtpmap:0 PCMU/8000 > >> a=rtpmap:8 PCMA/8000 > >> a=rtpmap:18 G729/8000 > >> a=rtpmap:9 G722/8000 > >> a=fmtp:101 0-15 > >> a=rtpmap:101 telephone-event/8000 > >> a=sendrecv > >> a=nortpproxy:yes > >> a=nortpproxy:yes > >> -----------LOG------------------ > >> DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found > valid > >> DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3 > >> DEBUG: rtpproxy [rtpproxy_funcs.c:148]: type <application/sdp> found > valid > >> DEBUG: rtpproxy [rtpproxy.c:2188]: proxy reply: 10068 192.168.1.3 > >> INFO: <script>: offer > >> -----------LOG------------------ > >> > >> -- > >> Regards, > >> > >> MingHon > > > > -- > Carsten Bock > http://www.ng-voice.com > mailto:cars...@ng-voice.com > > Schomburgstr. 80 > 22767 Hamburg > Germany > > Mobile +49 179 2021244 > Office +49 40 34927219 > Fax +49 40 34927220 > -- Regards, MingHon
_______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users